| 8432e1f4b84f79c4eea3a0820f4c4a83c267ef80 |
|
13-Jan-2016 |
marpan <marpan@google.com> |
Re-enable tests that failed under Linux_Msan.
Fixed in latest libvpx roll.
Keep EndToEndTest.TransportSeqNumOnAudioAndVideo disabled on
Win_DrMemory for now as it seems to time-out/too slow.
TBR=stefan@webrtc.org, kjellander@webrtc.org
BUG=webrtc:5402
NOTRY=True
Review URL: https://codereview.webrtc.org/1577313003
Cr-Commit-Position: refs/heads/master@{#11240}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e74eef19bd3f101208dc72b98038e42fc523a351 |
|
08-Jan-2016 |
stefan <stefan@webrtc.org> |
Add CreateSend/ReceiveTransport() methods to CallTest.
This allows the test to create its own transports if it, for instance, needs to do demuxing.
BUG=webrtc:5416
Review URL: https://codereview.webrtc.org/1573453002
Cr-Commit-Position: refs/heads/master@{#11187}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 9fea80f50daab46f20d4a6fc67b0144fbbbf56cd |
|
07-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Add audio streams to CallTest and a first A/V call test.
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.
Audio streams are using a fake audio device with file input.
The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.
R=pbos@webrtc.org
TBR=kjellander@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1542653002 .
Cr-Commit-Position: refs/heads/master@{#11171}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 44cc7950160aee01889a6b5a21a384b20cfb5532 |
|
07-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision 4df108a..2a70cb1 (367307:367468)
Mac 32-bit support has been gone in Chromium for a long time, but was
removed in https://codereview.chromium.org/1557823002. This called
for finally removing our Mac 32-bit builds, which was done in
http://crbug.com/574320.
Change log: https://chromium.googlesource.com/chromium/src/+log/4df108a..2a70cb1
Full diff: https://chromium.googlesource.com/chromium/src/+/4df108a..2a70cb1
Changed dependencies:
* src/third_party/libvpx_new/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/ecb8dff..a9dd8a7
* src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/aee1b12..225bfc3
DEPS diff: https://chromium.googlesource.com/chromium/src/+/4df108a..2a70cb1/DEPS
No update to Clang.
TBR=marpan@webrtc.org, stefan@webrtc.org,
BUG=webrtc:5401, webrtc:5402
NOTRY=True
Review URL: https://codereview.webrtc.org/1556273002
Cr-Commit-Position: refs/heads/master@{#11159}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 13f61dfea59a546e4e0081eb79e38c542ec51cf6 |
|
04-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Move fake-handle frame creation into test target.
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.
Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.
BUG=webrtc:5398
R=kjellander@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1554223002 .
Cr-Commit-Position: refs/heads/master@{#11149}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| cfb7f01fd627c666041355a69e70fa06ce149ece |
|
21-Dec-2015 |
honghaiz <honghaiz@webrtc.org> |
Disable VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly
due to flakiness on LinuxAsan.
BUG=webrtc:5382
TBR=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1541923003
Cr-Commit-Position: refs/heads/master@{#11109}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ff483617a4fdf282bb82d7f4ce15af3dbe305a4a |
|
21-Dec-2015 |
stefan <stefan@webrtc.org> |
Step 1 to prepare call_test.* for combined audio/video tests.
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.
No functional changes.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1537273003
Cr-Commit-Position: refs/heads/master@{#11101}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 38bb8ad1ca93d28d6f0055bb5663dff95d82a1e8 |
|
14-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Add test for verifying configured key frame interval for VP9.
BUG=
Review URL: https://codereview.webrtc.org/1498053002
Cr-Commit-Position: refs/heads/master@{#11004}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 5811a39f14fd77ebc0793ee93d03ee15a669bd8f |
|
10-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.
Does not modify test/channel_transport/.
BUG=
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1487893004 .
Cr-Commit-Position: refs/heads/master@{#10968}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d1590b2571c4cb33416e14c92e4f2dfed42ec3d4 |
|
09-Dec-2015 |
mflodman <mflodman@webrtc.org> |
Lint clean video/ and add lint presubmit check.
BUG=webrtc:5316
Review URL: https://codereview.webrtc.org/1507643004
Cr-Commit-Position: refs/heads/master@{#10953}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4654d204e42d00dea43ce1e5b2200063e8272c8b |
|
08-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Add test which verifies that the RTP header extensions are set correctly for FEC packets.
Also taking the opportunity to do a little bit of clean up.
BUG=webrtc:705
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1506863002 .
Cr-Commit-Position: refs/heads/master@{#10927}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ff24c04c7319478c71608974352e0d63e22f8589 |
|
04-Dec-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
Specify kf_min_dist to get correct key frame interval in svc mode.
Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).
BUG=chromium:500602
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1492633005 .
Cr-Commit-Position: refs/heads/master@{#10890}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e3384990ea2c2d1ab6ea4f00ac1f84c8322645d8 |
|
02-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Revert of Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. (patchset #18 id:580001 of https://codereview.webrtc.org/1437463002/ )
Reason for revert:
Breaks bots
Original issue's description:
> Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
>
> Specify kf_min_dist to get correct key frame interval in svc mode.
>
> BUG=chromium:500602
>
> Committed: https://crrev.com/43b48066a7d75bb051eea1e6f451147339cc98a6
> Cr-Commit-Position: refs/heads/master@{#10862}
TBR=pbos@webrtc.org,stefan@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500602
Review URL: https://codereview.webrtc.org/1492783002
Cr-Commit-Position: refs/heads/master@{#10863}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 43b48066a7d75bb051eea1e6f451147339cc98a6 |
|
02-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
Specify kf_min_dist to get correct key frame interval in svc mode.
BUG=chromium:500602
Review URL: https://codereview.webrtc.org/1437463002
Cr-Commit-Position: refs/heads/master@{#10862}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 12411ef40e08c5e28ccde54ab3418c96676ffcbc |
|
23-Nov-2015 |
pbos <pbos@webrtc.org> |
Move ThreadWrapper to ProcessThread in base.
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4b56904b70b2ad38c0790c0159819e89c05513b7 |
|
11-Nov-2015 |
stefan <stefan@webrtc.org> |
Fix race in VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.
BUG=webrtc:5194
Review URL: https://codereview.webrtc.org/1434963002
Cr-Commit-Position: refs/heads/master@{#10602}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| cfc319be1d6afec77bd41eeb70d3e7886dd524db |
|
10-Nov-2015 |
philipel <philipel@webrtc.org> |
Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..
(This time from my @webrtc account.)
Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1431283002
Cr-Commit-Position: refs/heads/master@{#10581}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 0be8f1d347bdb171462df89c2a4c69b3f3eb7519 |
|
10-Nov-2015 |
terelius <terelius@webrtc.org> |
Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1438543002
Cr-Commit-Position: refs/heads/master@{#10578}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 77ccfb4d16c148e61a316746bb5d9705e8b39f4a |
|
10-Nov-2015 |
philipel <philipel@webrtc.org> |
Work on flexible mode and screen sharing.
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.
Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1328113004
Cr-Commit-Position: refs/heads/master@{#10572}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 1295297153ff0487580faf821f24f09a7c16ce30 |
|
29-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Register header extensions in RtpRtcpObserver to avoid log spam.
BUG=webrtc:5118
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1416783006 .
Cr-Commit-Position: refs/heads/master@{#10450}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
|
27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 415d2cd7454d93b3727fce9147090a24e4c3ccba |
|
26-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for video.
BUG=webrtc:5118
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1415413004 .
Cr-Commit-Position: refs/heads/master@{#10403}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 22993e1a0c114122fc1b9de0fc74d4096ec868bd |
|
19-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.
BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1371043003
Cr-Commit-Position: refs/heads/master@{#10320}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| da903eaabbb6c6830efcafc3c2ade1d36f511e43 |
|
02-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1373903003
Cr-Commit-Position: refs/heads/master@{#10143}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 86fd9ed6f9e2a38aa343db8c62764659633231fa |
|
29-Sep-2015 |
sprang <sprang@webrtc.org> |
Set RtcpSender transport at construction.
BUG=
Review URL: https://codereview.webrtc.org/1365043002
Cr-Commit-Position: refs/heads/master@{#10106}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 5c389d3e09646c0e2ed76d5ccb37a3419a09eb6a |
|
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.
BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1227923005 .
Cr-Commit-Position: refs/heads/master@{#10073}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 586b19bdb615dde34cdcf107272d8857fe2f5631 |
|
18-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Enable probing with repeated payload packets by default.
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.
In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.
BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1327933003 .
Cr-Commit-Position: refs/heads/master@{#9984}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
|
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f325d2118c7d0e631121522be6ddd8eca8a215e2 |
|
08-Sep-2015 |
philipel <philipel@webrtc.org> |
Disable VideoSendStreamTest.VP9FlexMode.
Test is racy and fails on bots.
BUG=webrtc:4969
R=pbos@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1315803004 .
Cr-Commit-Position: refs/heads/master@{#9888}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 7fabd46a89675da596b28bb43c8fd3c561fbe85e |
|
03-Sep-2015 |
philipel <philipel@webrtc.org> |
Don't set V bit in flexible mode
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1291163007
Cr-Commit-Position: refs/heads/master@{#9848}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 0f9af0145683b9021ffe14da20c175bfa9db5cab |
|
01-Sep-2015 |
philipel <philipel@webrtc.org> |
Added send stream test case for VP9 header.
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1288363002
Cr-Commit-Position: refs/heads/master@{#9831}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d |
|
28-Aug-2015 |
solenberg <solenberg@webrtc.org> |
Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1273363005
Cr-Commit-Position: refs/heads/master@{#9807}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
|
03-Aug-2015 |
sprang <sprang@webrtc.org> |
Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d436298332c7a7ecb51241f3a66588539c2ece83 |
|
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove ResetStatistics from RTP feedback.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1213603002
Cr-Commit-Position: refs/heads/master@{#9548}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| eb66e800d1f5f74ab366715d2618fbede8cf3e12 |
|
05-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Re-land "Convert native handles to buffers before encoding."
This reverts commit a67675506c9057bd9ffd4d76aae8b743343d434d.
BUG=webrtc:4081
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1158273010
Cr-Commit-Position: refs/heads/master@{#9381}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 308d163c715df7b4348a1e00bf2a6761c0adb689 |
|
02-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Revert "Convert native handles to buffers before encoding."
This reverts commit a831dc3a7d10a1fbaa258ee6b1ca6cfc7e91c5ca to unblock
rolling into Chromium.
BUG=4081
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55549004
Cr-Commit-Position: refs/heads/master@{#9354}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| a831dc3a7d10a1fbaa258ee6b1ca6cfc7e91c5ca |
|
01-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Convert native handles to buffers before encoding.
Required to permit conversion of NV12 handles on iOS to I420 for VP8
software encoding, which blocks texture-based capture. This change
enforces that all texture-based input provides a method for converting
native handles to I420 if they are ever used with software encoders that
do not understand the native handles.
BUG=4081
R=emircan@chromium.org, glaznev@webrtc.org, hbos@webrtc.org, magjed@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50909005
Cr-Commit-Position: refs/heads/master@{#9347}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4765070b8d6f024509c717c04d9b708750666927 |
|
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.
Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.
BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52629004
Cr-Commit-Position: refs/heads/master@{#9339}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 242e22b055940be70b1df3031e2363b0d02397b2 |
|
11-May-2015 |
Erik Språng <sprang@webrtc.org> |
Refactor RTCP sender
The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but
it has quite a few ramifications. Notable changes:
* Removed the rtcpPacketTypeFlags bit vector and don't assume
RTCPPacketType values have a single unique bit set. This will allow
making this an enum class once rtcp_receiver has been overhauled.
* Flags are now stored in a map that is a member of the class. This
meant we could remove some bool flags (eg send_remb_) which was
previously masked into rtcpPacketTypeFlags and then masked out again
when testing if a remb packet should be sent.
* Make all build methods, eg. BuildREMB(), have the same signature.
An RtcpContext struct was introduced for this purpose. This allowed
the use of a map from RTCPPacketType to method pointer. Instead of
18 consecutive if-statements, there is now a single loop.
The context class also allowed some simplifications in the build
methods themselves.
* A few minor simplifications and cleanups.
The next step is to gradually replace the builder methods with the
builders from the new RtcpPacket classes.
BUG=2450
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48329004
Cr-Commit-Position: refs/heads/master@{#9166}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 01b488831bf7cb3276d8bdfbe0204dfbdbbba725 |
|
05-May-2015 |
Stefan Holmer <stefan@webrtc.org> |
Use padding to achieve bitrate probing if the initial key frame has too few packets.
BUG=4350
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44879004
Cr-Commit-Position: refs/heads/master@{#9134}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f2f828374c3ee1e1834c72bb27eaae88ef67bb40 |
|
01-May-2015 |
Peter Boström <pbos@webrtc.org> |
Use rtc::CriticalSection in webrtc/video/.
Removes heap allocation from CriticalSection creation.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50839004
Cr-Commit-Position: refs/heads/master@{#9126}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ff019b0b551888330b69d6323506eae710e1ab6d |
|
30-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Move rtc::AtomicOps to webrtc/base/atomicops.h.
Removes FixedSizeLockFreeQueue which isn't used anymore. This enabled
moving rtc::AtomicOps to webrtc/base/atomicops.h where they should be.
BUG=4330
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51789004
Cr-Commit-Position: refs/heads/master@{#9120}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 23fba1ffa0079f70744a83bcf4e85501dc226013 |
|
29-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51749004
Cr-Commit-Position: refs/heads/master@{#9114}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 143cec1cc68b9ba44f3ef4467f1422704f2395f0 |
|
28-Apr-2015 |
Erik Språng <sprang@google.com> |
Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.
BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46069004
Cr-Commit-Position: refs/heads/master@{#9093}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 9bfe3daf7349b62647997ced9389baa8ab043afe |
|
10-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Remove i420_video_frame.h header.
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 9526187dde1e93389b1d9077287eade974f9acfb |
|
10-Apr-2015 |
Erik Språng <sprang@google.com> |
Default enable abs send time bwe for CallTest
Using the single stream bwe is really bad for the screenshare
test case in particular, but would probably help in other
cases as well so enabling it by default in CallTest setup.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43089004
Cr-Commit-Position: refs/heads/master@{#8971}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 9b3f56ea055934a5d5416db0386c857494410acc |
|
09-Apr-2015 |
Per <perkj@chromium.org> |
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
This reverts commit e41d774c4d0a60066866fc2d0ae48dd0e839ff23.
Original code review: https://webrtc-codereview.appspot.com/43999004/
Reason for reland: There was nothing wrong with this cl as is, but it breaks chrome compatibility. We will now reland this and fix Chrome during roll.
Patset 1: Original cl.
Patchset 2: Removed more code that is no longer needed.
R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
BUG=1128
Review URL: https://webrtc-codereview.appspot.com/45049004
Cr-Commit-Position: refs/heads/master@{#8956}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e41d774c4d0a60066866fc2d0ae48dd0e839ff23 |
|
07-Apr-2015 |
Per <perkj@chromium.org> |
Revert "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."
This reverts commit 75db8612588b4fabdf1b05f4ab145f7737093b45.
Revert "Fix build breakage in WrappedI420Buffer::native_handle()"
This reverts commit 3211934ebf7cac3e6df2cb4aacb6e47cc1cffe2b.
Reason for revert: Breaks chrome build and tests on clank, See https://codereview.chromium.org/1067803002/
BUG=1128
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43079004
Cr-Commit-Position: refs/heads/master@{#8940}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 75db8612588b4fabdf1b05f4ab145f7737093b45 |
|
07-Apr-2015 |
Per <perkj@chromium.org> |
Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.
BUG=1128
R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43999004
Cr-Commit-Position: refs/heads/master@{#8932}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 53eda3dbd02a428178e7f9f40d2a4375c779cca8 |
|
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Add tests for r8811.
All these tests crashed before r8811. These tests should've been with
that change but r8811 was pushed in before to make bots green.
BUG=1788, 1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48669004
Cr-Commit-Position: refs/heads/master@{#8881}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e59041672283a28bde0b043c0c2bc198272f82e1 |
|
26-Mar-2015 |
Stefan Holmer <holmer@google.com> |
Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 2d2a30c2e22c16580193a8767bb4e7a2a3b30c00 |
|
24-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove I420VideoFrame::CloneFrame
This function is not needed anymore.
BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42899004
Cr-Commit-Position: refs/heads/master@{#8843}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 2b4ce3a501b8d679f84c1ad10317dea5c78fa595 |
|
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.
BUG=1756
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42879004
Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 143451d2590ef951f6e66a983a38a18fcd4c66a5 |
|
18-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Base start bitrate on last observed bitrate.
Instead of setting bitrates based on codec target settings (which may
have previously been capped by a codec max bitrate), fetch the last
bandwidth allocated for this channel. This fixes broken low start bitrates
due to QCIF being set as default codec in WebRtcVideoEngine2 which caps
the max bitrate to 200kbps.
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43789004
Cr-Commit-Position: refs/heads/master@{#8780}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| af612d5e0769571544952cbe55e675748afa9bdd |
|
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 93d9d6503e2bf2526af2b1c2cc46ef242b9843aa |
|
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45629004
Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 0d9bb8e499f52a53292fdb6dfa7dc956f6bff85b |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove the need for scoped_ptr<I420VideoFrame> in VieCapturer.
Remove the need for scoped_ptr<I420VideoFrame> in VieCapturer.
This adds the method I420VideoFrame::Reset and replace the use of scoped_ptr in ViECapturer.
Also, a unittest is added to check that ViECapturer does not retain a frame after it has been delivered.
BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43669004
Cr-Commit-Position: refs/heads/master@{#8678}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8678 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d7452a016812ab1de69c3d7a53caca5b06c64990 |
|
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| bcead305a2f27c30c72c6a3824fdf12f4b83c2eb |
|
06-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 45cdcce5f5c34d9321915473d8a0daafcf3abf78 |
|
06-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove TextureVideoFrame
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.
R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/40229004
Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 09c77b95bb62566be64da662f0b3b6a838ec6553 |
|
25-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add decoder-timing stats to VideoReceiveStream.
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 146742164664f9e35ed57575854e14f8011f02a8 |
|
23-Feb-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.
Only compare media bytes sent if number of sent packets in rtcp packet are equal to sent rtp packets.
BUG=4327
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34299004
Cr-Commit-Position: refs/heads/master@{#8454}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8454 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| fbcb5ceb166ed6b51be1da366817d64ecc86927a |
|
11-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove VideoSendStreamTest.ProducesStats.
This test is covered by EndToEndTests.GetStats and there's no need for a
duplicate test.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39049004
Cr-Commit-Position: refs/heads/master@{#8332}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8332 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| cfd82dfc1156f6610388bec0ebbdeacaf47e9719 |
|
22-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ce4e9a356200170abcdd44ff2af95f87a6781b8e |
|
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor some receive-side stats.
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 273a414b0ec2e58fdf3b817ad8b1a02f4ce15287 |
|
01-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report encoded frame size in VideoSendStream.
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d952c40c7e31c1603988c1f09ebfba9f17c6a866 |
|
27-Nov-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add receive bitrates to histogram stats:
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27189005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 008731868a09e2fe01da53733a612dc24761f791 |
|
25-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement settable min/start/max bitrates in Call.
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/26199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 049e4ece30b3901790949f9bbbeb5649a5c8932d |
|
20-Nov-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Change default values for CpuOveruseOptions.
Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85).
Moved DelayedEncoder to fake_encoder.h and added configuration for the delay.
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7722 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 49ff40e32e408bc77e8c9bec6090f6aa2e445173 |
|
13-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make SetREMBData accept vector of SSRCs.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 0b3d89b50059da945eee920a8c056c5ee8b5819d |
|
12-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7688 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
|
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| b7ed7799e77d3b315f5016951ecb90d18f10fdcb |
|
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement conference-mode temporal-layer screencast.
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ad3b5a5c16ff768def84138147d592ecb669a8cd |
|
24-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move min transmit bitrate to VideoEncoderConfig.
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 759982d357cb5d949b950218890b86c5026662eb |
|
22-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set number of temporal layers for VideoSendStream.
Introduces a mapping between EncoderConfig and VideoCodec. More
specifically it also removes an assert that there should be no set
temporal layers in the new API, which is wrong and was temporary.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7256 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| bbe0a8517d7f9da7aa779bff77cdbb70df358437 |
|
19-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4a6c5b3b019b23678b3b69f4a9d3a6042daebf89 |
|
15-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Re-enable video send stream tests for android.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7179 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 01581da71145d4b9504d12cfad0c988d1fc68654 |
|
04-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Fix audio/video sync when FEC is enabled.
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.
BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 6f729e8a74a4990ca2560607cbc9907cdfaf0401 |
|
02-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable video_engine_tests and webrtc_perf_tests on Android.
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| dde16f19e3ed36ca462f6404c40d5a9811f0ec37 |
|
06-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix some code styles.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22009004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 2f4b14e3f31b34a50310357c6c7be86c3bca1537 |
|
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTCP sender report send media bytes.
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4ef438e2defd6c46404f6b367287364cde66b7fb |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| bd9c0920ec75f97410a8753a91589bb3a70e9d1e |
|
10-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Skip encoding in fake VP8 encoder.
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 91f1752f2d34eee653f7693e09a485a8f5c50e1e |
|
10-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support VP8 encoder settings in VideoSendStream.
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 62bafae6618fe3aefbd18657062abc98a40c3375 |
|
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 161f8085000af32f094e0b903b7e2f7c19110b50 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add test for VideoEncoder setup/teardown.
Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().
BUG=2339
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 2bb1bdab8d11f5445693c028335fb3ace631f636 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| be9d2a45499d87f3b04e644fc173b0d997a9eeea |
|
30-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reserve RTP/RTCP modules in SetSSRC.
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 994d0b7229a18b255d81979c2bedaf8ecfae9bd7 |
|
27-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor Call-based tests.
Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.
BUG=3035
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f425b55eeb3711de323105b68559c6007829dc5f |
|
20-Jun-2014 |
wuchengli@chromium.org <wuchengli@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add tests of texture frames in video_send_stream_test.
Also fix a bug in ViEFrameProviderBase::DeliverFrame that
a texture frame was only delivered to the first callback.
BUG=chromium:362437
TEST=Run video engine test and webrtc call on CrOS.
R=kjellander@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, wuchengli@google.com
Review URL: https://webrtc-codereview.appspot.com/15789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6506 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| cb254aac3b18ac41ff175c816190390589182965 |
|
12-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable pacing by default and remove the option to disable it from the new API.
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 6ae48c660934784b4df56ab1ac99402ce3745e9f |
|
06-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VideoSendStream/VideoReceiveStream configs const.
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| caba2d2a370cb6b5e67c881ecfa57fdac7411de8 |
|
14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add DeliveryStatus enum to DeliverPacket().
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| de1429e9ad9a3a207ca191e1d748aa7271066860 |
|
28-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add thread annotations to Call API.
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| a5c8d2c9b39a2d20fead2147e60ed0cd6d62019c |
|
24-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename Start/Stop in Video{Send,Receive}Streams.
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.
R=mflodman@webrtc.org
BUG=3227
Review URL: https://webrtc-codereview.appspot.com/12329005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 44caf01c34d4fddec039f917c83fed7e0ce977b2 |
|
26-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-submit: rev5775
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 6cd201cf31dc8e50bf815139b0c9fdc83d3ba2bf |
|
25-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5775 "Modify bitrate controller to update bitrate based o..."
This triggered an occasional TSAN failure in
CallTest.ReceivesPliAndRecoversWithNack e.g.:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1444/steps/memory%20test%3A%20video_engine_tests/logs/stdio
I managed to reproduce this locally and verified that reverting this CL
corrected it.
> Modify bitrate controller to update bitrate based on process call and not
> only whenever a RTCP receiver block is received.
>
> Additionally:
> Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
>
> Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
>
> Did not touch decrease logic, however since it can be triggered more often it
> may decrease much faster and closer to the original written cap of once every
> 300ms + rtt.
>
> Note:
> rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
> bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
>
> BUG=3065
> R=stefan@webrtc.org, mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/10529004
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5785 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| da07737e68e23e283466ae21965e43edfe621a12 |
|
25-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5775 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 9af85c4ac22c65a323f42811b23ca14615f46481 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling SendsSetSimulcastSsrcs.
Disabling as bots are turning red. This should be because
VideoSendStream::ReconfigureVideoCodec caps video_codec.startBitrate to
max bitrates and as the start bitrate is just enough to transmit there
might be some rounding errors here causing the top stream not to be
sent. Since no REMB is received (send-side test) this remains as the
transmit bitrate.
I need some more time to figure out if this is the case so I'm disabling
these for now to avoid reverting the big CL. VideoSendStreams aren't
used in production yet.
TBR=mflodman@webrtc.org
BUG=3078
Review URL: https://webrtc-codereview.appspot.com/10229005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5727 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| add4073593a173ce2e41fe6afeda543020485c32 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable flaky CanSwitchToUseAllSsrcs.
Test flakes on bots, disabling while investigating.
R=minyue@webrtc.org
TBR=mflodman@webrtc.org
BUG=3078
Review URL: https://webrtc-codereview.appspot.com/10119006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5724 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 709e29742eb44a26bca3998d4c19797d6558775d |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Simplify pacer interface.
New interface uses two bitrates (max/min). The pace multiplier is also
removed from the interface and instead utilized outside. Min bitrate
will be filled with padding if there's not enough media to transmit.
Also fixes a bug in minimum transmission bitrate that made it ignore
REMBs. A regression test has been added to catch it.
BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5723 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f577ae9eac9822380ea6f0fb953cf383d0ec5374 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove internal codecs from VideoSendStream.
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.
For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.
Removes Call::GetVideoCodecs().
Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.
BUG=2854,2992
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ed8b2812659786106cd70592ed84f9f6475aaa7e |
|
18-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
The CL was reverted in r5712, due to bots going red. However, these bots
are unrelated to this CL.
Original description:
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was
that when the first non-padding packet was sent after the stream was
resumed, the statistics had not always been updated so that
stats.suspended was false. After seeing the first non-padding packet
after suspension, the test will now go into a state where it waits for
the statistics to be changed.
BUG=3068
R=pbos@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5713 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 12499ff20bbc8c23b0391905978cf5bdd50eb382 |
|
18-Mar-2014 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5711 "Fixing a flaky test in video_engine_tests"
> Fixing a flaky test in video_engine_tests
>
> VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed.
>
> BUG=3068
> R=pbos@webrtc.org
> TBR=stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/10069004
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5712 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d0f0c76cd9ea8fed1edce0f94832a6a659029c64 |
|
17-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing a flaky test in video_engine_tests
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed.
BUG=3068
R=pbos@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5711 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| b10363f3b63222b0f6ec7e916ef4ccac15d7205b |
|
13-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
This was originally committed as r5687, but reverted due to a flaky
test.
BUG=3040
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5695 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| be3947020382cc9733a9b53dff064f1353375bb5 |
|
11-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Routing SuspendChange to VideoSendStream::Stats"
The test VideoSendStreamTest.SuspendBelowMinBitrate seems flaky.
Reverting and investigating.
BUG=3040
Review URL: https://webrtc-codereview.appspot.com/9799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5681 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 1598b80f52bde9346f3eee20b08f51bcf5cfa245 |
|
11-Mar-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Routing SuspendChange to VideoSendStream::Stats
Also checking that the statistics are properly updated in
VideoSendStreamTest.SuspendBelowMinBitrate.
Adding a test to SendStatisticsProxyTest.
Checking callback status in rampup test, too.
BUG=2457
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 60ad5fdadf486ad2d516f1d9baeeed7e0fee67f9 |
|
06-Mar-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Potential deadlock in VideoSendStreamTest::ProducesStats
VideoSendStream::GetStats() should not be called by
RtpRtcpObserver::OnSendRtcp(), as at this stage that thread will still
hold internal send locks.
Use an event and signal the test thread to call GetStats() instead.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5648 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 346094cb01ef2ffbf0398f465d61c9a4f77b465c |
|
18-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Incorrect overhead calculation when using FEC + RTP extension headers.
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| d9b9560ee50c236efcb690ee479021b415f7dfd4 |
|
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Drop early packets when not sending in TransportAdapter.
Particularly, suppress periodic RTCP packets before
VideoSendStream.StartSending() or VideoReceiveStream.StartReceiving() have been called, respectively.
RTCP packets are sent periodically, by the Process thread, for every ViE channel even those not sending.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5438 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| c279a5d72c885b1a1737018ee26dc7c0475a38bf |
|
24-Jan-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up RTX in VideoReceiveStream.
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.
BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e7223e7795b8581575fcaf58be971f2ed8f8af2e |
|
23-Jan-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set NACKed packet to -1 in TestNackRetransmission.
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.
BUG=2830
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| f777cf254787adc944d3b641f6c525329bfe2a20 |
|
10-Jan-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Permitting double start/stopping of streams.
It doesn't make too much sense to hard enforce that the user keeps track
of which streams are started and which are not.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5363 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ccd42840bcee8db145be91b3308912a24f710a6f |
|
07-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up statistics in video send stream of new video engine api
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5559006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 54ae4ffb9e235a9742e2b11298327e02d870571c |
|
19-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTCP statistics.
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.
TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.
BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 052fa6243a175f1f31410b4d2223b7ad48cbca40 |
|
17-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Stop transport in test SuspendBelowMinBitrate.
Avoids race when packets are still left in the network while the Call is
being destroyed.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/6009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5307 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 5ab756703ea32f2c2ff9878d6eae628c7380bc14 |
|
16-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r5294 to re-roll r5293.
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 41e2615e020311172b937f527c13d9e090437eca |
|
15-Dec-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 341e91441aaa9c2c5a638082c3ee4530aa21612c |
|
14-Dec-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 724947b8efa44d15d699b471020005450590f5b6 |
|
11-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SwapFrame() to VideoSendStreamInput.
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 8b8819262f4ec2333656aa6eb0b76e7a893b683a |
|
10-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improve VideoSendStreamTest::MaxPacketSize
This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/, but was reverted because of flakiness. This new issue will correct that.
Patch Set 1 contains the code that was submitted in 4849004.
BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 797522f9f260d8945265f9531c4dfaf89f5b8ecc |
|
06-Dec-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
> Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
>
> BUG=2428
> R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4849004
It caused a failure in video_engine_tests on the Linux Tsan bot.
TBR=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5240 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 7104fc1906cfe20f143bd8d15a31810b026dc7d0 |
|
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| b613b5ab2b03041942f04fd892e2ad5a4f9de027 |
|
03-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set local SSRC for VideoReceiveStream.
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.
BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| e1fc3f22ea0d2f2b66e0ff65c175aa22bdb502d7 |
|
28-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable check for all sent SSRCs being valid.
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.
This part should be added back during work on issue 1695.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 13d38a13e3a6b55d0a95bbac32963014bd0a2002 |
|
28-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set up SSRCs correctly after switching codec.
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.
No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4ab4fc00449448570dcb1ddf9e89752d5a4890b4 |
|
25-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add test for automatically disabling padding when no video is being captured.
BUG=2648
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5169 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 331d4402fca65fcccf0f4c93958d79f47fe58165 |
|
21-Nov-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Connect pacer/padding to SuspendBelowMinBitrate
The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.
Had to change the unit test to make it aware of the padding packets.
BUG=2606
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 2c46f8d854c1fc3e10f8151ee5109923287aee8b |
|
21-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename DestroyStream methods to include Video.
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ce90eff3456fdfc6f15b3639c96593ac3924ac66 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename RTP-extension constants.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 53c85735256dc7d540deb0a5e2bbb2f2821c4bd4 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename video streams' start/stop methods.
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 5a63655ab0de18bd2fa376ba4774eab3f3bc9fb2 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename Call::Create{Receive,Send}Stream().
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| ce8e0936d988a6d3fa075ab9ff954b690d503718 |
|
18-Nov-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename AutoMute to SuspendBelowMinBitrate
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 4cfa6050f638952f080b671e3c74bc8fff0ba9ce |
|
15-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix breakage after introducing new test.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3899005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 69969e2e2f0420df2765ab72d8e6f96d6d9d5d9c |
|
15-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improve Call tests for RTX.
Also does some refactoring to reuse RtpRtcpObserver.
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 6488761f2e6ce7b977bbc14bc7b91933527d633a |
|
14-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| def22b455b818217201a110fb9b4ef65bc18378e |
|
29-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Stop DirectTransports in VideoSendStreamTests.
Prevents racy packet delivery during or after Call destruction.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3099005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|
| 16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
|
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream_tests.cc
|